Job Description
Walk-in Interview – VoIP EngineerAbout startelelogic
startelelogic is an
AI-powered Omnichannel Customer Experience (CX) platform
that helps businesses deliver seamless and intelligent interactions across
voice, chat, email, WhatsApp, social media, and more . Our platform brings together
Conversational AI, Voice AI, AI Agents, automation, analytics, and human-assisted workflows
to simplify customer support, sales, and engagement. We are looking for a motivated
VoIP Engineer with 1+ year of hands-on experience
in VoIP and SIP technologies to join our telecom engineering team. The ideal candidate should have practical knowledge of
SIP, VoIP call flows, Linux, FreeSWITCH, and/or Kamailio , along with basic programming and scripting skills. The candidate will work on developing, configuring, troubleshooting, and maintaining reliable, carrier-grade VoIP systems.
Key Responsibilities Configure, troubleshoot, and maintain
FreeSWITCH and Kamailio
platforms. Understand and troubleshoot
SIP signaling and VoIP call flows . Analyze SIP messages such as
INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, 200 OK, 4xx, and 5xx
responses. Troubleshoot common VoIP issues, including: One-way or no audio Call failures SIP registration issues DTMF problems Codec negotiation issues NAT-related issues Call routing failures Work with
SIP trunks, extensions, gateways, DIDs, and VoIP routing . Monitor and troubleshoot production VoIP services. Use tools such as
sngrep, tcpdump, Wireshark, netstat/ss , and other Linux utilities for debugging. Work with
FreeSWITCH dial plans, XML configurations, and SIP profiles . Assist with
Kamailio routing and configuration . Perform SIP trunk testing, call-flow analysis, and connectivity troubleshooting. Write or modify basic scripts using
Python, Lua, Perl, or Bash . Collaborate with senior engineers to implement new VoIP features and telecom integrations. Monitor system performance and help maintain service availability. Document configurations, troubleshooting procedures, and technical changes. Required Skills 1+ year of hands-on experience
in VoIP/SIP engineering. Basic to intermediate understanding of
VoIP and SIP . Hands-on experience with
FreeSWITCH and/or Kamailio . Good understanding of
Linux . Ability to read and troubleshoot
SIP traces and call flows . Understanding of
TCP/IP, UDP, DNS, NAT, and basic networking concepts . Basic knowledge of
RTP and media flow . Basic programming or scripting knowledge in
Python, Lua, Perl, Bash, or Golang . Basic knowledge of
C/C++
is an advantage. Strong troubleshooting, analytical, and problem-solving skills. Ability to work effectively in a production telecom environment. Good to Have Experience with
OpenSIPS
or other SIP proxy technologies. Knowledge of
RTP, SRTP, and media handling . Familiarity with
SIP over UDP, TCP, and TLS . Knowledge of codecs such as
G.711, G.729, Opus, and GSM . Exposure to
REST APIs and telecom integrations . Basic understanding of databases such as
MySQL, PostgreSQL, or Redis . Experience with
Git and Linux-based development environments . Exposure to
Docker, Kubernetes, or cloud-based VoIP deployments
is a plus. Education B.Tech/B.E. in Computer Science, Electronics, Electronics & Communication, or Telecommunications
is required. BCA, MCA, diploma courses, or other non-engineering qualifications will not be considered. Candidates should have a strong academic foundation in computer networks, telecommunications, Linux, and VoIP/SIP technologies . Fresh
B.Tech/B.E. graduates
with strong VoIP/SIP fundamentals and relevant practical projects may also be considered. Walk-in Interview Details Date:
September 5, 2026,
Saturday Time:
11:00 AM – 3:00 PM Position:
VoIP Engineer Experience:
1+ Year Open Positions:
5 Employment Type:
Full-Time Work Mode:
On-site
Interview Venue First Floor, NX ONE, T1, 117 & 118 Greater Noida West, Ghaziabad Ithaira, Uttar Pradesh – 201318
What You Will Work On As a VoIP Engineer at
startelelogic , you will work on real-world
telecom and carrier-grade communication systems , including
VoIP infrastructure, SIP routing, FreeSWITCH, Kamailio, SIP trunks, carrier connectivity, call routing, troubleshooting, and AI-powered customer experience platforms . This role provides an opportunity to gain hands-on experience across
VoIP, CCaaS, CPaaS, AI-powered communications, and cloud communication technologies
while working with a growing telecom technology company.
⚠️ Important Please bring an updated copy of your
resume/CV
and be prepared for a
technical discussion/interview focused on VoIP, SIP, Linux, FreeSWITCH, Kamailio, and networking fundamentals .
AI-powered Omnichannel Customer Experience (CX) platform
that helps businesses deliver seamless and intelligent interactions across
voice, chat, email, WhatsApp, social media, and more . Our platform brings together
Conversational AI, Voice AI, AI Agents, automation, analytics, and human-assisted workflows
to simplify customer support, sales, and engagement. We are looking for a motivated
VoIP Engineer with 1+ year of hands-on experience
in VoIP and SIP technologies to join our telecom engineering team. The ideal candidate should have practical knowledge of
SIP, VoIP call flows, Linux, FreeSWITCH, and/or Kamailio , along with basic programming and scripting skills. The candidate will work on developing, configuring, troubleshooting, and maintaining reliable, carrier-grade VoIP systems.
Key Responsibilities Configure, troubleshoot, and maintain
FreeSWITCH and Kamailio
platforms. Understand and troubleshoot
SIP signaling and VoIP call flows . Analyze SIP messages such as
INVITE, ACK, BYE, CANCEL, REGISTER, OPTIONS, 200 OK, 4xx, and 5xx
responses. Troubleshoot common VoIP issues, including: One-way or no audio Call failures SIP registration issues DTMF problems Codec negotiation issues NAT-related issues Call routing failures Work with
SIP trunks, extensions, gateways, DIDs, and VoIP routing . Monitor and troubleshoot production VoIP services. Use tools such as
sngrep, tcpdump, Wireshark, netstat/ss , and other Linux utilities for debugging. Work with
FreeSWITCH dial plans, XML configurations, and SIP profiles . Assist with
Kamailio routing and configuration . Perform SIP trunk testing, call-flow analysis, and connectivity troubleshooting. Write or modify basic scripts using
Python, Lua, Perl, or Bash . Collaborate with senior engineers to implement new VoIP features and telecom integrations. Monitor system performance and help maintain service availability. Document configurations, troubleshooting procedures, and technical changes. Required Skills 1+ year of hands-on experience
in VoIP/SIP engineering. Basic to intermediate understanding of
VoIP and SIP . Hands-on experience with
FreeSWITCH and/or Kamailio . Good understanding of
Linux . Ability to read and troubleshoot
SIP traces and call flows . Understanding of
TCP/IP, UDP, DNS, NAT, and basic networking concepts . Basic knowledge of
RTP and media flow . Basic programming or scripting knowledge in
Python, Lua, Perl, Bash, or Golang . Basic knowledge of
C/C++
is an advantage. Strong troubleshooting, analytical, and problem-solving skills. Ability to work effectively in a production telecom environment. Good to Have Experience with
OpenSIPS
or other SIP proxy technologies. Knowledge of
RTP, SRTP, and media handling . Familiarity with
SIP over UDP, TCP, and TLS . Knowledge of codecs such as
G.711, G.729, Opus, and GSM . Exposure to
REST APIs and telecom integrations . Basic understanding of databases such as
MySQL, PostgreSQL, or Redis . Experience with
Git and Linux-based development environments . Exposure to
Docker, Kubernetes, or cloud-based VoIP deployments
is a plus. Education B.Tech/B.E. in Computer Science, Electronics, Electronics & Communication, or Telecommunications
is required. BCA, MCA, diploma courses, or other non-engineering qualifications will not be considered. Candidates should have a strong academic foundation in computer networks, telecommunications, Linux, and VoIP/SIP technologies . Fresh
B.Tech/B.E. graduates
with strong VoIP/SIP fundamentals and relevant practical projects may also be considered. Walk-in Interview Details Date:
September 5, 2026,
Saturday Time:
11:00 AM – 3:00 PM Position:
VoIP Engineer Experience:
1+ Year Open Positions:
5 Employment Type:
Full-Time Work Mode:
On-site
Interview Venue First Floor, NX ONE, T1, 117 & 118 Greater Noida West, Ghaziabad Ithaira, Uttar Pradesh – 201318
What You Will Work On As a VoIP Engineer at
startelelogic , you will work on real-world
telecom and carrier-grade communication systems , including
VoIP infrastructure, SIP routing, FreeSWITCH, Kamailio, SIP trunks, carrier connectivity, call routing, troubleshooting, and AI-powered customer experience platforms . This role provides an opportunity to gain hands-on experience across
VoIP, CCaaS, CPaaS, AI-powered communications, and cloud communication technologies
while working with a growing telecom technology company.
⚠️ Important Please bring an updated copy of your
resume/CV
and be prepared for a
technical discussion/interview focused on VoIP, SIP, Linux, FreeSWITCH, Kamailio, and networking fundamentals .
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